1. Field of the Invention
The present invention relates to a voice data communication system, and in particular to a voice data communication system accommodating an IP telephone network in which a VoIP (voice over IP) technology enabling a voice call through a LAN/WAN (Local Area Network/Wide Area Network) is used.
2. Description of the Related Art
FIG. 13 shows an arrangement of a prior art voice data communication system. The voice data communication system accommodating an IP telephone network which applies thereto an already-existing line switching network and a VoIP technology, shown in this prior art example, is composed of a switchboard 1, a LAN or a WAN (hereinafter, generally referred to as LAN for simplifying descriptions) 2, a gateway (hereinafter, occasionally abbreviated as GW) 3 which interworks a protocol on the LAN or WAN 2 with a protocol for a line switching, IP extension terminals 4a and 4b (hereinafter, occasionally represented by a reference numeral “4”), and a general extension terminal 5.
The IP extension terminal 4 is an apparatus, connected to the LAN 2 instead of an existing telephone line, having functions of encoding a voice signal and making it a voice data packet (UDP packet) for the transmission, or of decoding a received voice data packet for the reproduction.
Since the above-mentioned voice data communication system includes a configuration in which the IP extension terminal 4a or 4b communicates with the general extension terminal 5 by telephone, a decision processing of a called extension terminal and a line switching processing are all performed by the switchboard 1.
Therefore, even if it is a communication between the IP extention terminals 4a and 4b connected to each other with the LAN 2, the voice data packet cannot be directly transmitted/received between the IP extention terminals, so that the voice data packet must be transmitted through the gateway 3 and the switchboard 1.
Hereinafter, details of the connecting operation between the IP extension terminals 4a and 4b accommodated in the gateway 3 will be described referring to FIG. 13.
The switchboard 1 manages which of the gateway 3 accommodates the IP extension terminals 4a and 4b, and determines a called extension terminal based on call control signal information from a calling extension terminal. In case this called extension terminal is an IP extension terminal, the switchboard 1 performs a called processing to the gateway 3.
The gateway 3 performs a processing of tying a logical line of the LAN 2 on the side of the IP extension terminals 4a and 4b with a TDM (Time Division Multiplexing) line on the switchboard 1 side. There are a call control signal route R1, a media control signal route R2, and a voice data route R3 for a logical line on the LAN 2 side, as shown in FIG. 13.
[Establishment of Call Control Signal Route R1]
The switchboard 1 and the gateway 3 establish routes R12 and R13 within the call control signal route R1 by using an existing, individual signal line etc. established in the TDM line.
The gateway 3 and the IP extension terminals 4a, 4b are connected through the LAN 2, so that the call control is carried out by a TCP connection with a high reliability. The IP extension terminals 4a and 4b have preliminarily known destination information (IP address+TCP port) for a call control signal route establishment of the gateway 3 in which the IP extension terminals 4a and 4b themselves are accommodated, and the gateway 3 has preliminarily known destination information (IP address+TCP port) of the accommodating IP extension terminals.
The calling IP extension terminal 4a establishes a TCP connection for the gateway 3. The IP extension terminal 4a edits called No. information of the IP extension terminal 4b to a session initialize message to be transmitted to the gateway 3 through the TCP connection (R11 in FIG. 13).
The gateway 3 transmits the received session initialize message to the call control signal route R12.
The switchboard 1 having received the session initialize message through the call control signal route R12 recognizes that the called party IP extension terminal 4b is accommodated in the gateway 3 from the called party No. information, and transmits the session initialize message to the gateway 3 (R13 in FIG. 13).
The gateway 3 establishes the TCP connection of a call control signal route R14 for the called IP extension terminal 4b. 
Hereafter, the IP extension terminals 4a and 4b transmit/receive a call control signal message through the TCP connection of the LAN line and the call control signal route R1 on the TDM line, thereby leading to a call state.
[Establishment of Voice Data Route R3]
Routes R32 and R33 forming the voice data route R3 on the TDM line side are determined by the call control message between the switchboard 1 and the gateway 3.
Voice data routes R31 and R34 on the LAN 2 side are established by using the media control signal route R2. The reason is that a voice data compression-coding rule used is required to be determined since voice data compression-coding is generally performed for the voice data packet on LAN 2, and that the destination information (IP address+UDP port No.) to transmit/receive the voice data packet is variable. Hereinafter, how to establish the media control signal route R2 will be described.
The IP extension terminal 4a establishes a media controlling TCP connection between the IP extension terminal 4a and the gateway 3, determines the voice data compression-coding rule by using the TCP connection, and notifies the destination information (IP address+UDP port No.) for a voice data packet reception to each other.
The IP extension terminals 4a and 4b or the gateway 3 transmits the voice data packet on the destination information of the destination notified by the media control signal routes R21 and R22. Thus, a bi-directional voice data packet route R3 is established between the IP extension terminals 4a and 4b. 
The IP extension terminals 4a and 4b compression-code the inputted voice signal, and make the encoded voice data a UDP packet (voice data packet) to be transmitted. Also, the IP extension terminals 4a and 4b decompression-decode the voice data of the received voice data packet, and reproduce the voice signal.
The gateway 3 transmits the PCM data in which the voice data of the received voice data packet are decompression-decoded, to the voice data route R32 of the TDM line. Also, the PCM data received from the voice data route R33 of the TDM line are compression-coded, and the encoded voice data are made the UDP packet to be transmitted.
The switchboard 1 inserts the PCM data from the calling party channel of the voice data route R32 into the called party channel of the voice data route R33, and inserts the PCM data from the called party channel into the calling party channel.
Thus, there have been following problems in the prior art since the voice data route R3 passes through the gateway 3 and the switchboard 1 in spite of a call between the IP extension terminals 4a and 4b. 
The first problem is that a delay time increases due to an assembly/disassembly of the voice data into the IP packet in the gateway 3.
When the voice data transformed to the IP packet on nonguaranteed QoS (Quality of Service) network (LAN/WAN) are transmitted, it is necessary to absorb a dispersion (fluctuation) of the transmission delay in the network on the called party, so that the called party usually accumulates the voice data packet for a predetermined time to be decoded.
Delay a speaker feels for a voice packet's arrival can be decreased if the accumulating time is shortened. However in that case, a packet delayed by more than the accumulating time cannot be decoded, resulting in deteriorated voice quality. Namely, a delay time for absorbing the fluctuation must be accepted to some extent as a result of a trade-off with the voice quality.
A transmission line of an IP base is originally inferior to a TDM transmission line and an ATM transmission line in a delay time, so that a simple addition of a delay time of fluctuation absorption causes a worse influence upon the call quality.
The second problem is that the voice quality is deteriorated since the compression-coding and the decompression-decoding of the voice data are performed when passing through the gateway 3.
A voice waveform generally distorts by a quantization noise in a voice data compression-coding. The noise is little in the inputted voice signal, when the distortion is allowable. However, when the voice signal to which the quantization noise is added is inputted, the voice quality is largely deteriorated. Increased frequency of the compression-coding and decompression-decoding of the voice data further deteriorates the voice quality.
The third problem is that the voice data packet of the twice amount is transmitted/received compared with the case where the voice data packet is directly transmitted/received between the IP extension terminals, thereby decreasing a band use efficiency of the network.